In the field of broadcasting radio and television programs, the sound part of the program is the subject of various processings and manipulations of the signal. These processes are designed, on the one hand, to prevent any exceeding of thresholds regulated in terms of electric level, and, on the other hand, to obtain a subjective form of the sound texture in order to give a sound identity to the broadcast programs and/or a higher sound power impression than that of the competition.
The different processes for modifying an audiofrequency signal are numerous and varied, and the range of effects, such as the pallet of optimizing these effects, are developing rapidly.
Basically, modifying a sound signal requires knowing, above all, the exact characteristics of the original sound signal.
Therefore, it is indispensable to qualify this original audiofrequency signal by evaluations, measurements, analyses, to obtain the maximum characteristics of the signal at a given instant to decide on the nature of the modifications that must be made to the signal at this instant in order to achieve the objective set by the conversion process (for example, higher sound power impression, etc.).
The quality of the processing (effectiveness, absence of any audible distortions, low latency time, etc.) is therefore linked to two essential parameters:                The quality of the measurement of the original signal,        The quality of the processings carried out after the measurement and depending on the results of same.        
Many audiofrequency processing devices designed for radio or television (regardless of the broadcasting mode: Hertzian, satellite, cable, internet network or others) are known in the prior art. These have proved to be limited in the development of their functionalities and in the improvement of the quality of the processings because of the mediocre quality of the measurements supposed to rigorously represent the original signal. The existing audio signal processing devices have therefore reached a technological stage limited by the performance of the process for measuring the original audio signal.
The techniques used by these devices to ensure the measurement of an audio signal, in order to obtain representative data, are commonly based on an analog evaluation of the signal (especially by successive mean value methods).
Sophisticated processes have been used to overcome the existence of these imprecisions in the measurement of the original audio signal. However, they do not make it possible to obtain reliable and stable results on the entire audio spectrum or on the entire dynamic range of the signal, resulting in the appearance of audible distortions and of “false friends” (artifacts) triggering unexpected processings of the signal.
Other prior-art audio signal processing devices use a technique in which the original signal is rather simply evaluated, and the “corrective” processes are applied to the processed signal. Thus, if on such a signal pattern it is known that the measurement performed upstream is affected by a nonlinearity, a corrective is used on the processed signal to limit the amplitude of the processing affected by the poor starting evaluation of the original audio signal.
However, this method, adapted in always identical particular cases, fails when the sound programs are highly varied (classical music then modern music, for example). Therefore, it is not polyvalent.
Finally, in “high end” audio processing equipment, processes are known which combine the two methods according to essentially empirical weightings, which prove to be extremely delicate to use during operations of installing and adjusting the equipment.
It is seen that the prior-art solutions are not satisfactory because of existing errors on the measurements of the original signal, which are never corrected perfectly.